Complete and systematic signal processing reference on VoIP Voice and Fax signal processing Covers several aspects of deployment and connecting practical aspects with the module and the algorithms. Shows the reader how to troubleshoot the problems discussed in the book.

Informationen zum Autor Sivannarayana Nagireddi, PhD , is the architect of VoIP solutions at Ikanos Communications, Inc., and leads the DSP and VoIP team. Dr. Nagireddi and his team developed complete VoIP solutions including signal processing algorithms for voice-enabled residential gateway processors, which have been deployed by telecommunications providers. He is a member of the IEEE, a Fellow of IETE-India, and reviewer for the journal Medical Engineering & Physics. Klappentext A complete and systematic treatment of signal processing for VoIP voice and fax This book presents a consolidated view and basic approach to signal processing for VoIP voice and fax solutions. It provides readers with complete coverage of the topic, from how things work in voice and fax modules, to signal processing aspects, implementation, and testing. Beginning with an overview of VoIP infrastructure, interfaces, and signals, the book systematically covers: Voice compression Packet loss concealment techniques DTMF detection, generation, and rejection Wideband voice modules operation VoIP Voice-Network bit rate calculations VoIP voice testing Fax over IP and modem over IP Country deviations of PSTN mapped to VoIP VoIP on different processors and architectures Generic VAD-CNG for waveform codecs Echo cancellation Caller ID features in VoIP Packetization-RTP, RTCP, and jitter buffer Clock sources for VoIP applications Fax operation on PSTN, modulations, and fax messages Fax over IP payload formats and bit rate calculations Voice packets jitter with large data packets VoIP voice quality Over 100 questions and answers on voice and more than seventy questions and answers on fax are provided at the back of the book to reinforce the topics covered throughout the text. Additionally, several clarification, interpretation, and discussion sections are included in selected chapters to aide in readers' comprehension. VoIP Voice and Fax Signal Processing is an indispensable resource for professional electrical engineers, voice and fax solution developers, product and deployment support teams, quality assurance and test engineers, and computer engineers. It also serves as a valuable textbook for graduate-level students in electrical engineering and computer engineering courses. Zusammenfassung Complete and systematic signal processing reference on VoIP Voice and Fax signal processing Covers several aspects of deployment and connecting practical aspects with the module and the algorithms. Shows the reader how to troubleshoot the problems discussed in the book. Inhaltsverzeichnis Acknowledgments xix About the Author xxi Preface xxiii Glossary xxvii 1 PSTN Basic Infrastructure, Interfaces, and Signals 1 1.1 PSTN CO and DLC 2 1.2 PSTN User Interfaces 3 1.3 Data Services on Telephone Lines 7 1.4 Power Levels and Digital Quantization for G.711 A-Law 9 1.5 Significance of Power Levels on Listening 11 1.6 TR-57, IEEE-743, and TIA Standards Overview 13 2 VoIP Overview and Infrastructure 19 2.1 PSTN and VoIP 20 2.2 Typical VoIP Deployment Example 25 2.3 Network and Acoustic Interfaces for VoIP 26 2.4 VoIP Systems Working Principles 27 2.5 VoIP Signaling 41 3 Voice Compression 49 3.1 Compression Codecs 50 3.2 G.711 Compression 50 3.3 Speech Redundancies and Compression 60 3.4 G.726 or ADPCM Compression 60 3.5 Wideband Voice 62 3.6 G.729 Family of Low-Bit-Rate Codecs 63 3.7 Miscellaneous Narrow and Wideband Codecs 67 3.8 Codecs and Overload Levels 70 3.9 Voice Quality of Codecs 70 3.10 C-Source Code for Codecs 74 ...

Autorentext

Sivannarayana Nagireddi, PhD, is the architect of VoIP solutions at Ikanos Communications, Inc., and leads the DSP and VoIP team. Dr. Nagireddi and his team developed complete VoIP solutions including signal processing algorithms for voice-enabled residential gateway processors, which have been deployed by telecommunications providers. He is a member of the IEEE, a Fellow of IETE-India, and reviewer for the journal Medical Engineering & Physics.

Klappentext

A complete and systematic treatment of signal processing for VoIP voice and fax This book presents a consolidated view and basic approach to signal processing for VoIP voice and fax solutions. It provides readers with complete coverage of the topic, from how things work in voice and fax modules, to signal processing aspects, implementation, and testing. Beginning with an overview of VoIP infrastructure, interfaces, and signals, the book systematically covers:

• Voice compression

• Packet loss concealment techniques

• DTMF detection, generation, and rejection

• Wideband voice modules operation

• VoIP Voice-Network bit rate calculations

• VoIP voice testing

• Fax over IP and modem over IP

• Country deviations of PSTN mapped to VoIP

• VoIP on different processors and architectures

• Generic VAD-CNG for waveform codecs

• Echo cancellation

• Caller ID features in VoIP

• Packetization-RTP, RTCP, and jitter buffer

• Clock sources for VoIP applications

• Fax operation on PSTN, modulations, and fax messages

• Fax over IP payload formats and bit rate calculations

• Voice packets jitter with large data packets

• VoIP voice quality

Over 100 questions and answers on voice and more than seventy questions and answers on fax are provided at the back of the book to reinforce the topics covered throughout the text. Additionally, several clarification, interpretation, and discussion sections are included in selected chapters to aide in readers' comprehension.

VoIP Voice and Fax Signal Processing is an indispensable resource for professional electrical engineers, voice and fax solution developers, product and deployment support teams, quality assurance and test engineers, and computer engineers. It also serves as a valuable textbook for graduate-level students in electrical engineering and computer engineering courses.

Inhalt

Acknowledgments xix

About the Author xxi

Preface xxiii

Glossary xxvii

1 PSTN Basic Infrastructure, Interfaces, and Signals 1

1.1 PSTN CO and DLC 2

1.2 PSTN User Interfaces 3

1.3 Data Services on Telephone Lines 7

1.4 Power Levels and Digital Quantization for G.711 A-Law 9

1.5 Significance of Power Levels on Listening 11

1.6 TR-57, IEEE-743, and TIA Standards Overview 13

2 VoIP Overview and Infrastructure 19

2.1 PSTN and VoIP 20

2.2 Typical VoIP Deployment Example 25

2.3 Network and Acoustic Interfaces for VoIP 26

2.4 VoIP Systems Working Principles 27

2.5 VoIP Signaling 41

3 Voice Compression 49

3.1 Compression Codecs 50

3.2 G.711 Compression 50

3.3 Speech Redundancies and Compression 60

3.4 G.726 or ADPCM Compression 60

3.5 Wideband Voice 62

3.6 G.729 Family of Low-Bit-Rate Codecs 63

3.7 Miscellaneous Narrow and Wideband Codecs 67

3.8 Codecs and Overload Levels 70

3.9 Voice Quality of Codecs 70

3.10 C-Source Code for Codecs 74

3.11 Codecs in VoIP Deployment 74

4 Generic VADCNG for Waveform codecs 76

4.1 VADCNG and Codecs 77

4.2 Generic VADCNG Functionality 78

4.3 Comfort Noise Payload Format 78

4.4 G.711 Appendix II VADCNG Algorithm 80

4.5 Power-Based VADCNG 83

4.6 VADCNG in Low-Bit-Rate Codecs 85

4.7 Miscellaneous Aspects of VADCNG 86

4.8 Summary on VADCNG 89

5 Packet Loss Concealment Techniques 91

5.1 Packet Loss Concealment Overview 91

5.2 Packet Loss Concealment Techniques 92

5.3 Transmitter- and Receiver-Based Techniques 94

5.4 Decoder-Only Based PLC Techniques 99

5.5 PLC Techniques Description 101

5.6 PLC for Low-Bit-Rate Codecs 108

5.7 PLC Testing 110

5.8 PLC Summary and Discussion 111

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